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Created Speex payloader and depayloader; Optimize G711 payloader to use adapter and send packets until MTU size.

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Original commit message from CVS:
Created Speex payloader and depayloader; Optimize G711 payloader to use adapter and send packets until MTU size.
This commit is contained in:
Edgard Lima 2005-11-17 18:23:23 +00:00
parent 42c5075f17
commit 5ae66f78c5
17 changed files with 1018 additions and 63 deletions
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17
ChangeLog
View file

@ -1,3 +1,20 @@
2005-11-17 Edgard Lima <edgard.lima@indt.org.br>
* gst/rtp/Makefile.am
* gst/rtp/gstrtp.c
* gst/rtp/gstrtpg711enc.c: (gst_rtpg711enc_src_template),
(gst_rtpg711enc_class_init), (gst_rtpg711enc_init),
(gst_rtpg711enc_finalize), (gst_rtpg711enc_setcaps),
(gst_rtpg711enc_flush), (gst_rtpg711enc_handle_buffer):
* gst/rtp/gstrtpg711enc.h:
* gst/rtp/gstrtpg711dec.c: (gst_rtpg711dec_sink_template):
* gst/rtp/gstrtpspeexenc.c:
* gst/rtp/gstrtpspeexenc.h:
* gst/rtp/gstrtpspeexdec.c:
* gst/rtp/gstrtpspeexdec.h:
Created Speex payloader and depayloader; Optimize G711 payloader to
use adapter and send packets until MTU size.
2005-11-16 Wim Taymans <wim@fluendo.com>
* check/elements/matroskamux.c: (setup_src_pad), (setup_sink_pad):

10
gst/rtp/Makefile.am
View file

@ -16,8 +16,10 @@ libgstrtp_la_SOURCES = \
gstrtph263enc.c \
gstasteriskh263.c \
gstrtpmp4venc.c \
gstrtpmp4vdec.c
gstrtpmp4vdec.c \
gstrtpspeexenc.c \
gstrtpspeexdec.c
#gstrtpL16enc.c gstrtpL16parse.c gstrtpgsmenc.c gstrtpgsmparse.c
if HAVE_WINSOCK2_H
@ -48,4 +50,6 @@ noinst_HEADERS = gstrtpL16enc.h \
gstrtpmp4vdec.h \
gstrtpdec.h \
gstrtph263enc.h \
gstasteriskh263.h
gstasteriskh263.h \
gstrtpspeexenc.h \
gstrtpspeexdec.h

8
gst/rtp/gstrtp.c
View file

@ -36,6 +36,8 @@
#include "gstasteriskh263.h"
#include "gstrtpmp4venc.h"
#include "gstrtpmp4vdec.h"
#include "gstrtpspeexenc.h"
#include "gstrtpspeexdec.h"
static gboolean
plugin_init (GstPlugin * plugin)
@ -85,6 +87,12 @@ plugin_init (GstPlugin * plugin)
if (!gst_rtpmp4vdec_plugin_init (plugin))
return FALSE;
if (!gst_rtpspeexenc_plugin_init (plugin))
return FALSE;
if (!gst_rtpspeexdec_plugin_init (plugin))
return FALSE;
return TRUE;
}

4
gst/rtp/gstrtpg711dec.c
View file

@ -48,12 +48,12 @@ static GstStaticPadTemplate gst_rtpg711dec_sink_template =
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) 0, "
"payload = (int) [ 0, 255 ], "
"clock-rate = (int) 8000, "
"encoding-name = (string) \"PCMU\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) 8, "
"payload = (int) [ 0, 255 ], "
"clock-rate = (int) 8000, " "encoding-name = (string) \"PCMA\"")
);

4
gst/rtp/gstrtpg711depay.c
View file

@ -48,12 +48,12 @@ static GstStaticPadTemplate gst_rtpg711dec_sink_template =
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) 0, "
"payload = (int) [ 0, 255 ], "
"clock-rate = (int) 8000, "
"encoding-name = (string) \"PCMU\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) 8, "
"payload = (int) [ 0, 255 ], "
"clock-rate = (int) 8000, " "encoding-name = (string) \"PCMA\"")
);

117
gst/rtp/gstrtpg711enc.c
View file

@ -45,12 +45,12 @@ static GstStaticPadTemplate gst_rtpg711enc_src_template =
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) 0, "
"payload = (int) " GST_RTP_PAYLOAD_PCMU_STRING ", "
"clock-rate = (int) 8000, "
"encoding-name = (string) \"PCMU\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) 8, "
"payload = (int) " GST_RTP_PAYLOAD_PCMA_STRING ", "
"clock-rate = (int) 8000, " "encoding-name = (string) \"PCMA\"")
);
@ -58,6 +58,7 @@ static gboolean gst_rtpg711enc_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtpg711enc_handle_buffer (GstBaseRTPPayload * payload,
GstBuffer * buffer);
static void gst_rtpg711enc_finalize (GObject * object);
GST_BOILERPLATE (GstRtpG711Enc, gst_rtpg711enc, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
@ -86,6 +87,7 @@ gst_rtpg711enc_class_init (GstRtpG711EncClass * klass)
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
gobject_class->finalize = gst_rtpg711enc_finalize;
gstbasertppayload_class->set_caps = gst_rtpg711enc_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtpg711enc_handle_buffer;
@ -94,9 +96,23 @@ gst_rtpg711enc_class_init (GstRtpG711EncClass * klass)
static void
gst_rtpg711enc_init (GstRtpG711Enc * rtpg711enc, GstRtpG711EncClass * klass)
{
rtpg711enc->adapter = gst_adapter_new ();
GST_BASE_RTP_PAYLOAD (rtpg711enc)->clock_rate = 8000;
}
static void
gst_rtpg711enc_finalize (GObject * object)
{
GstRtpG711Enc *rtpg711enc;
rtpg711enc = GST_RTP_G711_ENC (object);
g_object_unref (rtpg711enc->adapter);
rtpg711enc->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtpg711enc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
@ -109,9 +125,11 @@ gst_rtpg711enc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
stname = gst_structure_get_name (structure);
if (0 == strcmp ("audio/x-mulaw", stname)) {
gst_basertppayload_set_options (payload, "audio", TRUE, "PCMU", 8000);
payload->pt = GST_RTP_PAYLOAD_PCMU;
gst_basertppayload_set_options (payload, "audio", FALSE, "PCMU", 8000);
} else if (0 == strcmp ("audio/x-alaw", stname)) {
gst_basertppayload_set_options (payload, "audio", TRUE, "PCMA", 8000);
payload->pt = GST_RTP_PAYLOAD_PCMA;
gst_basertppayload_set_options (payload, "audio", FALSE, "PCMA", 8000);
} else {
return FALSE;
}
@ -121,42 +139,89 @@ gst_rtpg711enc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
return TRUE;
}
static GstFlowReturn
gst_rtpg711enc_flush (GstRtpG711Enc * rtpg711enc)
{
guint avail;
GstBuffer *outbuf;
GstFlowReturn ret;
/* the data available in the adapter is either smaller
* than the MTU or bigger. In the case it is smaller, the complete
* adapter contents can be put in one packet. */
avail = gst_adapter_available (rtpg711enc->adapter);
ret = GST_FLOW_OK;
while (avail > 0) {
guint towrite;
guint8 *payload;
guint8 *data;
guint payload_len;
guint packet_len;
/* this will be the total lenght of the packet */
packet_len = gst_rtpbuffer_calc_packet_len (avail, 0, 0);
/* fill one MTU or all available bytes */
towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpg711enc));
/* this is the payload length */
payload_len = gst_rtpbuffer_calc_payload_len (towrite, 0, 0);
/* create buffer to hold the payload */
outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
/* copy payload */
gst_rtpbuffer_set_payload_type (outbuf,
GST_BASE_RTP_PAYLOAD_PT (rtpg711enc));
payload = gst_rtpbuffer_get_payload (outbuf);
data = (guint8 *) gst_adapter_peek (rtpg711enc->adapter, payload_len);
memcpy (payload, data, payload_len);
gst_adapter_flush (rtpg711enc->adapter, payload_len);
avail -= payload_len;
GST_BUFFER_TIMESTAMP (outbuf) = rtpg711enc->first_ts;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpg711enc), outbuf);
}
return ret;
}
static GstFlowReturn
gst_rtpg711enc_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpG711Enc *rtpg711enc;
guint size, payload_len;
GstBuffer *outbuf;
guint8 *payload, *data;
GstClockTime timestamp;
guint size, packet_len, avail;
GstFlowReturn ret;
GstClockTime duration;
rtpg711enc = GST_RTP_G711_ENC (basepayload);
size = GST_BUFFER_SIZE (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_TIMESTAMP (buffer);
/* FIXME, only one G711 frame per RTP packet for now */
payload_len = size;
avail = gst_adapter_available (rtpg711enc->adapter);
if (avail == 0) {
rtpg711enc->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtpg711enc->duration = 0;
}
outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
/* FIXME, assert for now */
g_assert (payload_len <= GST_BASE_RTP_PAYLOAD_MTU (rtpg711enc));
/* get packet length of data and see if we exceeded MTU. */
packet_len = gst_rtpbuffer_calc_packet_len (avail + size, 0, 0);
/* copy timestamp */
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
/* get payload */
payload = gst_rtpbuffer_get_payload (outbuf);
/* if this buffer is going to overflow the packet, flush what we
* have. */
if (gst_basertppayload_is_filled (basepayload,
packet_len, rtpg711enc->duration + duration)) {
ret = gst_rtpg711enc_flush (rtpg711enc);
rtpg711enc->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtpg711enc->duration = 0;
} else {
ret = GST_FLOW_OK;
}
data = GST_BUFFER_DATA (buffer);
/* copy data in payload */
memcpy (&payload[0], data, size);
gst_buffer_unref (buffer);
ret = gst_basertppayload_push (basepayload, outbuf);
gst_adapter_push (rtpg711enc->adapter, buffer);
rtpg711enc->duration += duration;
return ret;
}

7
gst/rtp/gstrtpg711enc.h
View file

@ -18,6 +18,7 @@
#include <gst/gst.h>
#include <gst/rtp/gstbasertppayload.h>
#include <gst/base/gstadapter.h>
G_BEGIN_DECLS
@ -38,8 +39,10 @@ typedef struct _GstRtpG711EncClass GstRtpG711EncClass;
struct _GstRtpG711Enc
{
GstBaseRTPPayload payload;
gint frequency;
GstAdapter *adapter;
GstClockTime first_ts;
GstClockTime duration;
};
struct _GstRtpG711EncClass

117
gst/rtp/gstrtpg711pay.c
View file

@ -45,12 +45,12 @@ static GstStaticPadTemplate gst_rtpg711enc_src_template =
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) 0, "
"payload = (int) " GST_RTP_PAYLOAD_PCMU_STRING ", "
"clock-rate = (int) 8000, "
"encoding-name = (string) \"PCMU\"; "
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) 8, "
"payload = (int) " GST_RTP_PAYLOAD_PCMA_STRING ", "
"clock-rate = (int) 8000, " "encoding-name = (string) \"PCMA\"")
);
@ -58,6 +58,7 @@ static gboolean gst_rtpg711enc_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtpg711enc_handle_buffer (GstBaseRTPPayload * payload,
GstBuffer * buffer);
static void gst_rtpg711enc_finalize (GObject * object);
GST_BOILERPLATE (GstRtpG711Enc, gst_rtpg711enc, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
@ -86,6 +87,7 @@ gst_rtpg711enc_class_init (GstRtpG711EncClass * klass)
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
gobject_class->finalize = gst_rtpg711enc_finalize;
gstbasertppayload_class->set_caps = gst_rtpg711enc_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtpg711enc_handle_buffer;
@ -94,9 +96,23 @@ gst_rtpg711enc_class_init (GstRtpG711EncClass * klass)
static void
gst_rtpg711enc_init (GstRtpG711Enc * rtpg711enc, GstRtpG711EncClass * klass)
{
rtpg711enc->adapter = gst_adapter_new ();
GST_BASE_RTP_PAYLOAD (rtpg711enc)->clock_rate = 8000;
}
static void
gst_rtpg711enc_finalize (GObject * object)
{
GstRtpG711Enc *rtpg711enc;
rtpg711enc = GST_RTP_G711_ENC (object);
g_object_unref (rtpg711enc->adapter);
rtpg711enc->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtpg711enc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
@ -109,9 +125,11 @@ gst_rtpg711enc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
stname = gst_structure_get_name (structure);
if (0 == strcmp ("audio/x-mulaw", stname)) {
gst_basertppayload_set_options (payload, "audio", TRUE, "PCMU", 8000);
payload->pt = GST_RTP_PAYLOAD_PCMU;
gst_basertppayload_set_options (payload, "audio", FALSE, "PCMU", 8000);
} else if (0 == strcmp ("audio/x-alaw", stname)) {
gst_basertppayload_set_options (payload, "audio", TRUE, "PCMA", 8000);
payload->pt = GST_RTP_PAYLOAD_PCMA;
gst_basertppayload_set_options (payload, "audio", FALSE, "PCMA", 8000);
} else {
return FALSE;
}
@ -121,42 +139,89 @@ gst_rtpg711enc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
return TRUE;
}
static GstFlowReturn
gst_rtpg711enc_flush (GstRtpG711Enc * rtpg711enc)
{
guint avail;
GstBuffer *outbuf;
GstFlowReturn ret;
/* the data available in the adapter is either smaller
* than the MTU or bigger. In the case it is smaller, the complete
* adapter contents can be put in one packet. */
avail = gst_adapter_available (rtpg711enc->adapter);
ret = GST_FLOW_OK;
while (avail > 0) {
guint towrite;
guint8 *payload;
guint8 *data;
guint payload_len;
guint packet_len;
/* this will be the total lenght of the packet */
packet_len = gst_rtpbuffer_calc_packet_len (avail, 0, 0);
/* fill one MTU or all available bytes */
towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpg711enc));
/* this is the payload length */
payload_len = gst_rtpbuffer_calc_payload_len (towrite, 0, 0);
/* create buffer to hold the payload */
outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
/* copy payload */
gst_rtpbuffer_set_payload_type (outbuf,
GST_BASE_RTP_PAYLOAD_PT (rtpg711enc));
payload = gst_rtpbuffer_get_payload (outbuf);
data = (guint8 *) gst_adapter_peek (rtpg711enc->adapter, payload_len);
memcpy (payload, data, payload_len);
gst_adapter_flush (rtpg711enc->adapter, payload_len);
avail -= payload_len;
GST_BUFFER_TIMESTAMP (outbuf) = rtpg711enc->first_ts;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpg711enc), outbuf);
}
return ret;
}
static GstFlowReturn
gst_rtpg711enc_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpG711Enc *rtpg711enc;
guint size, payload_len;
GstBuffer *outbuf;
guint8 *payload, *data;
GstClockTime timestamp;
guint size, packet_len, avail;
GstFlowReturn ret;
GstClockTime duration;
rtpg711enc = GST_RTP_G711_ENC (basepayload);
size = GST_BUFFER_SIZE (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_TIMESTAMP (buffer);
/* FIXME, only one G711 frame per RTP packet for now */
payload_len = size;
avail = gst_adapter_available (rtpg711enc->adapter);
if (avail == 0) {
rtpg711enc->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtpg711enc->duration = 0;
}
outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
/* FIXME, assert for now */
g_assert (payload_len <= GST_BASE_RTP_PAYLOAD_MTU (rtpg711enc));
/* get packet length of data and see if we exceeded MTU. */
packet_len = gst_rtpbuffer_calc_packet_len (avail + size, 0, 0);
/* copy timestamp */
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
/* get payload */
payload = gst_rtpbuffer_get_payload (outbuf);
/* if this buffer is going to overflow the packet, flush what we
* have. */
if (gst_basertppayload_is_filled (basepayload,
packet_len, rtpg711enc->duration + duration)) {
ret = gst_rtpg711enc_flush (rtpg711enc);
rtpg711enc->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtpg711enc->duration = 0;
} else {
ret = GST_FLOW_OK;
}
data = GST_BUFFER_DATA (buffer);
/* copy data in payload */
memcpy (&payload[0], data, size);
gst_buffer_unref (buffer);
ret = gst_basertppayload_push (basepayload, outbuf);
gst_adapter_push (rtpg711enc->adapter, buffer);
rtpg711enc->duration += duration;
return ret;
}

7
gst/rtp/gstrtpg711pay.h
View file

@ -18,6 +18,7 @@
#include <gst/gst.h>
#include <gst/rtp/gstbasertppayload.h>
#include <gst/base/gstadapter.h>
G_BEGIN_DECLS
@ -38,8 +39,10 @@ typedef struct _GstRtpG711EncClass GstRtpG711EncClass;
struct _GstRtpG711Enc
{
GstBaseRTPPayload payload;
gint frequency;
GstAdapter *adapter;
GstClockTime first_ts;
GstClockTime duration;
};
struct _GstRtpG711EncClass

143
gst/rtp/gstrtpspeexdec.c Normal file
View file

@ -0,0 +1,143 @@
/* GStreamer
* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpspeexdec.h"
/* elementfactory information */
static GstElementDetails gst_rtp_speexdec_details = {
"RTP packet parser",
"Codec/Parser/Network",
"Extracts Speex audio from RTP packets",
"Edgard Lima <edgard.lima@indt.org.br>"
};
/* RtpSPEEXDec signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0
};
static GstStaticPadTemplate gst_rtpspeexdec_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) [ 96, 127 ], "
"clock-rate = (int) [6000, 48000], "
"encoding-name = (string) \"speex\", "
"encoding-params = (string) \"1\"")
);
static GstStaticPadTemplate gst_rtpspeexdec_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-speex")
);
static GstBuffer *gst_rtpspeexdec_process (GstBaseRTPDepayload * depayload,
GstBuffer * buf);
static gboolean gst_rtpspeexdec_setcaps (GstBaseRTPDepayload * depayload,
GstCaps * caps);
GST_BOILERPLATE (GstRtpSPEEXDec, gst_rtpspeexdec, GstBaseRTPDepayload,
GST_TYPE_BASE_RTP_DEPAYLOAD);
static void
gst_rtpspeexdec_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpspeexdec_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpspeexdec_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_speexdec_details);
}
static void
gst_rtpspeexdec_class_init (GstRtpSPEEXDecClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_DEPAYLOAD);
gstbasertpdepayload_class->process = gst_rtpspeexdec_process;
gstbasertpdepayload_class->set_caps = gst_rtpspeexdec_setcaps;
}
static void
gst_rtpspeexdec_init (GstRtpSPEEXDec * rtpspeexdec, GstRtpSPEEXDecClass * klass)
{
GST_BASE_RTP_DEPAYLOAD (rtpspeexdec)->clock_rate = 8000;
}
static gboolean
gst_rtpspeexdec_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstCaps *srccaps;
gboolean ret;
srccaps = gst_static_pad_template_get_caps (&gst_rtpspeexdec_src_template);
ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
return ret;
}
static GstBuffer *
gst_rtpspeexdec_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstBuffer *outbuf = NULL;
gint payload_len;
guint8 *payload;
GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
GST_BUFFER_SIZE (buf),
gst_rtpbuffer_get_marker (buf),
gst_rtpbuffer_get_timestamp (buf), gst_rtpbuffer_get_seq (buf));
payload_len = gst_rtpbuffer_get_payload_len (buf);
payload = gst_rtpbuffer_get_payload (buf);
outbuf = gst_buffer_new_and_alloc (payload_len);
memcpy (GST_BUFFER_DATA (outbuf), payload, payload_len);
return outbuf;
}
gboolean
gst_rtpspeexdec_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpspeexdec",
GST_RANK_NONE, GST_TYPE_RTP_SPEEX_DEC);
}

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/* GStreamer
* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more
*/
#ifndef __GST_RTP_SPEEX_DEC_H__
#define __GST_RTP_SPEEX_DEC_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertpdepayload.h>
G_BEGIN_DECLS
typedef struct _GstRtpSPEEXDec GstRtpSPEEXDec;
typedef struct _GstRtpSPEEXDecClass GstRtpSPEEXDecClass;
#define GST_TYPE_RTP_SPEEX_DEC \
(gst_rtpspeexdec_get_type())
#define GST_RTP_SPEEX_DEC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SPEEX_DEC,GstRtpSPEEXDec))
#define GST_RTP_SPEEX_DEC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SPEEX_DEC,GstRtpSPEEXDec))
#define GST_IS_RTP_SPEEX_DEC(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SPEEX_DEC))
#define GST_IS_RTP_SPEEX_DEC_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SPEEX_DEC))
struct _GstRtpSPEEXDec
{
GstBaseRTPDepayload depayload;
};
struct _GstRtpSPEEXDecClass
{
GstBaseRTPDepayloadClass parent_class;
};
gboolean gst_rtpspeexdec_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_SPEEX_DEC_H__ */

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/* GStreamer
* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpspeexdec.h"
/* elementfactory information */
static GstElementDetails gst_rtp_speexdec_details = {
"RTP packet parser",
"Codec/Parser/Network",
"Extracts Speex audio from RTP packets",
"Edgard Lima <edgard.lima@indt.org.br>"
};
/* RtpSPEEXDec signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0
};
static GstStaticPadTemplate gst_rtpspeexdec_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) [ 96, 127 ], "
"clock-rate = (int) [6000, 48000], "
"encoding-name = (string) \"speex\", "
"encoding-params = (string) \"1\"")
);
static GstStaticPadTemplate gst_rtpspeexdec_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-speex")
);
static GstBuffer *gst_rtpspeexdec_process (GstBaseRTPDepayload * depayload,
GstBuffer * buf);
static gboolean gst_rtpspeexdec_setcaps (GstBaseRTPDepayload * depayload,
GstCaps * caps);
GST_BOILERPLATE (GstRtpSPEEXDec, gst_rtpspeexdec, GstBaseRTPDepayload,
GST_TYPE_BASE_RTP_DEPAYLOAD);
static void
gst_rtpspeexdec_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpspeexdec_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpspeexdec_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_speexdec_details);
}
static void
gst_rtpspeexdec_class_init (GstRtpSPEEXDecClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_DEPAYLOAD);
gstbasertpdepayload_class->process = gst_rtpspeexdec_process;
gstbasertpdepayload_class->set_caps = gst_rtpspeexdec_setcaps;
}
static void
gst_rtpspeexdec_init (GstRtpSPEEXDec * rtpspeexdec, GstRtpSPEEXDecClass * klass)
{
GST_BASE_RTP_DEPAYLOAD (rtpspeexdec)->clock_rate = 8000;
}
static gboolean
gst_rtpspeexdec_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstCaps *srccaps;
gboolean ret;
srccaps = gst_static_pad_template_get_caps (&gst_rtpspeexdec_src_template);
ret = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
return ret;
}
static GstBuffer *
gst_rtpspeexdec_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstBuffer *outbuf = NULL;
gint payload_len;
guint8 *payload;
GST_DEBUG ("process : got %d bytes, mark %d ts %u seqn %d",
GST_BUFFER_SIZE (buf),
gst_rtpbuffer_get_marker (buf),
gst_rtpbuffer_get_timestamp (buf), gst_rtpbuffer_get_seq (buf));
payload_len = gst_rtpbuffer_get_payload_len (buf);
payload = gst_rtpbuffer_get_payload (buf);
outbuf = gst_buffer_new_and_alloc (payload_len);
memcpy (GST_BUFFER_DATA (outbuf), payload, payload_len);
return outbuf;
}
gboolean
gst_rtpspeexdec_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpspeexdec",
GST_RANK_NONE, GST_TYPE_RTP_SPEEX_DEC);
}

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/* GStreamer
* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more
*/
#ifndef __GST_RTP_SPEEX_DEC_H__
#define __GST_RTP_SPEEX_DEC_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertpdepayload.h>
G_BEGIN_DECLS
typedef struct _GstRtpSPEEXDec GstRtpSPEEXDec;
typedef struct _GstRtpSPEEXDecClass GstRtpSPEEXDecClass;
#define GST_TYPE_RTP_SPEEX_DEC \
(gst_rtpspeexdec_get_type())
#define GST_RTP_SPEEX_DEC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SPEEX_DEC,GstRtpSPEEXDec))
#define GST_RTP_SPEEX_DEC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SPEEX_DEC,GstRtpSPEEXDec))
#define GST_IS_RTP_SPEEX_DEC(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SPEEX_DEC))
#define GST_IS_RTP_SPEEX_DEC_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SPEEX_DEC))
struct _GstRtpSPEEXDec
{
GstBaseRTPDepayload depayload;
};
struct _GstRtpSPEEXDecClass
{
GstBaseRTPDepayloadClass parent_class;
};
gboolean gst_rtpspeexdec_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_SPEEX_DEC_H__ */

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/* GStreamer
* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpspeexenc.h"
/* elementfactory information */
static GstElementDetails gst_rtpspeexenc_details = {
"RTP packet parser",
"Codec/Encoder/Network",
"Encodes Speex audio into a RTP packet",
"Edgard Lima <edgard.lima@indt.org.br>"
};
static GstStaticPadTemplate gst_rtpspeexenc_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-speex")
);
static GstStaticPadTemplate gst_rtpspeexenc_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) 110, " /* guaranties compatibility with Linphone
Could be [96,127] See page 34 at http://www.ietf.org/rfc/rfc3551.txt */
"clock-rate = (int) [6000, 48000], "
"encoding-name = (string) \"speex\", "
"encoding-params = (string) \"1\"")
);
static gboolean gst_rtpspeexenc_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtpspeexenc_handle_buffer (GstBaseRTPPayload * payload,
GstBuffer * buffer);
GST_BOILERPLATE (GstRtpSPEEXEnc, gst_rtpspeexenc, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
static void
gst_rtpspeexenc_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpspeexenc_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpspeexenc_src_template));
gst_element_class_set_details (element_class, &gst_rtpspeexenc_details);
}
static void
gst_rtpspeexenc_class_init (GstRtpSPEEXEncClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
gstbasertppayload_class->set_caps = gst_rtpspeexenc_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtpspeexenc_handle_buffer;
}
static void
gst_rtpspeexenc_init (GstRtpSPEEXEnc * rtpspeexenc, GstRtpSPEEXEncClass * klass)
{
GST_BASE_RTP_PAYLOAD (rtpspeexenc)->clock_rate = 8000;
GST_BASE_RTP_PAYLOAD_PT (rtpspeexenc) = 110; /* Create String */
}
static gboolean
gst_rtpspeexenc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
gst_basertppayload_set_options (payload, "audio", FALSE, "speex", 8000);
gst_basertppayload_set_outcaps (payload, NULL);
return TRUE;
}
static GstFlowReturn
gst_rtpspeexenc_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpSPEEXEnc *rtpspeexenc;
guint size, payload_len;
GstBuffer *outbuf;
guint8 *payload, *data;
GstClockTime timestamp;
GstFlowReturn ret;
rtpspeexenc = GST_RTP_SPEEX_ENC (basepayload);
size = GST_BUFFER_SIZE (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
/* FIXME, only one SPEEX frame per RTP packet for now */
payload_len = size;
outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
/* FIXME, assert for now */
g_assert (payload_len <= GST_BASE_RTP_PAYLOAD_MTU (rtpspeexenc));
/* copy timestamp */
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
/* get payload */
payload = gst_rtpbuffer_get_payload (outbuf);
data = GST_BUFFER_DATA (buffer);
/* copy data in payload */
memcpy (&payload[0], data, size);
gst_buffer_unref (buffer);
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
}
gboolean
gst_rtpspeexenc_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpspeexenc",
GST_RANK_NONE, GST_TYPE_RTP_SPEEX_ENC);
}

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/* GStreamer
* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more
*/
#ifndef __GST_RTP_SPEEX_ENC_H__
#define __GST_RTP_SPEEX_ENC_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertppayload.h>
G_BEGIN_DECLS
typedef struct _GstRtpSPEEXEnc GstRtpSPEEXEnc;
typedef struct _GstRtpSPEEXEncClass GstRtpSPEEXEncClass;
#define GST_TYPE_RTP_SPEEX_ENC \
(gst_rtpspeexenc_get_type())
#define GST_RTP_SPEEX_ENC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SPEEX_ENC,GstRtpSPEEXEnc))
#define GST_RTP_SPEEX_ENC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SPEEX_ENC,GstRtpSPEEXEnc))
#define GST_IS_RTP_SPEEX_ENC(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SPEEX_ENC))
#define GST_IS_RTP_SPEEX_ENC_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SPEEX_ENC))
struct _GstRtpSPEEXEnc
{
GstBaseRTPPayload payload;
};
struct _GstRtpSPEEXEncClass
{
GstBaseRTPPayloadClass parent_class;
};
gboolean gst_rtpspeexenc_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_SPEEX_ENC_H__ */

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/* GStreamer
* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpspeexenc.h"
/* elementfactory information */
static GstElementDetails gst_rtpspeexenc_details = {
"RTP packet parser",
"Codec/Encoder/Network",
"Encodes Speex audio into a RTP packet",
"Edgard Lima <edgard.lima@indt.org.br>"
};
static GstStaticPadTemplate gst_rtpspeexenc_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-speex")
);
static GstStaticPadTemplate gst_rtpspeexenc_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) 110, " /* guaranties compatibility with Linphone
Could be [96,127] See page 34 at http://www.ietf.org/rfc/rfc3551.txt */
"clock-rate = (int) [6000, 48000], "
"encoding-name = (string) \"speex\", "
"encoding-params = (string) \"1\"")
);
static gboolean gst_rtpspeexenc_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtpspeexenc_handle_buffer (GstBaseRTPPayload * payload,
GstBuffer * buffer);
GST_BOILERPLATE (GstRtpSPEEXEnc, gst_rtpspeexenc, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
static void
gst_rtpspeexenc_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpspeexenc_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtpspeexenc_src_template));
gst_element_class_set_details (element_class, &gst_rtpspeexenc_details);
}
static void
gst_rtpspeexenc_class_init (GstRtpSPEEXEncClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_BASE_RTP_PAYLOAD);
gstbasertppayload_class->set_caps = gst_rtpspeexenc_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtpspeexenc_handle_buffer;
}
static void
gst_rtpspeexenc_init (GstRtpSPEEXEnc * rtpspeexenc, GstRtpSPEEXEncClass * klass)
{
GST_BASE_RTP_PAYLOAD (rtpspeexenc)->clock_rate = 8000;
GST_BASE_RTP_PAYLOAD_PT (rtpspeexenc) = 110; /* Create String */
}
static gboolean
gst_rtpspeexenc_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
gst_basertppayload_set_options (payload, "audio", FALSE, "speex", 8000);
gst_basertppayload_set_outcaps (payload, NULL);
return TRUE;
}
static GstFlowReturn
gst_rtpspeexenc_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpSPEEXEnc *rtpspeexenc;
guint size, payload_len;
GstBuffer *outbuf;
guint8 *payload, *data;
GstClockTime timestamp;
GstFlowReturn ret;
rtpspeexenc = GST_RTP_SPEEX_ENC (basepayload);
size = GST_BUFFER_SIZE (buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
/* FIXME, only one SPEEX frame per RTP packet for now */
payload_len = size;
outbuf = gst_rtpbuffer_new_allocate (payload_len, 0, 0);
/* FIXME, assert for now */
g_assert (payload_len <= GST_BASE_RTP_PAYLOAD_MTU (rtpspeexenc));
/* copy timestamp */
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
/* get payload */
payload = gst_rtpbuffer_get_payload (outbuf);
data = GST_BUFFER_DATA (buffer);
/* copy data in payload */
memcpy (&payload[0], data, size);
gst_buffer_unref (buffer);
ret = gst_basertppayload_push (basepayload, outbuf);
return ret;
}
gboolean
gst_rtpspeexenc_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpspeexenc",
GST_RANK_NONE, GST_TYPE_RTP_SPEEX_ENC);
}

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/* GStreamer
* Copyright (C) <2005> Edgard Lima <edgard.lima@indt.org.br>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more
*/
#ifndef __GST_RTP_SPEEX_ENC_H__
#define __GST_RTP_SPEEX_ENC_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertppayload.h>
G_BEGIN_DECLS
typedef struct _GstRtpSPEEXEnc GstRtpSPEEXEnc;
typedef struct _GstRtpSPEEXEncClass GstRtpSPEEXEncClass;
#define GST_TYPE_RTP_SPEEX_ENC \
(gst_rtpspeexenc_get_type())
#define GST_RTP_SPEEX_ENC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SPEEX_ENC,GstRtpSPEEXEnc))
#define GST_RTP_SPEEX_ENC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SPEEX_ENC,GstRtpSPEEXEnc))
#define GST_IS_RTP_SPEEX_ENC(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SPEEX_ENC))
#define GST_IS_RTP_SPEEX_ENC_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SPEEX_ENC))
struct _GstRtpSPEEXEnc
{
GstBaseRTPPayload payload;
};
struct _GstRtpSPEEXEncClass
{
GstBaseRTPPayloadClass parent_class;
};
gboolean gst_rtpspeexenc_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_SPEEX_ENC_H__ */